1. Field of the Invention
The present invention relates generally to an audio decoder. In particular, the present invention relates to a method and circuit for implementing a dual-mode audio decoder which performs an inverse modified discrete cosine transform (IMDCT) on a signal encoded using the Moving Picture Experts Group (MPEG) standard and the Dolby.RTM. AC-3 standard. The IMDCT transform is performed using a common Fast Fourier Transform (FFT) circuit. The audio decoder reduces the size of necessary memory by reducing the number of IMDCT outputs used in windowing and by utilizing the properties of the IMDSCT outputs.
2. Description of the Related Art
As the processing of digital audio has increased in the video and multimedia fields, so has the demand for effective compression algorithms. Effective compression algorithms are necessary because digital audio occupies a considerable portion of the signal bandwidth. Representative compression algorithms are MPEG and Dolby AC-3. Those of skill in the art will appreciate that the MPEG and AC-33 algorithms are well known, evolving standards. Accordingly, reference herein to these standards will be understood to mean the standards as they existed at the time of the earliest effective filing date of the present application, and as they have evolved to date, and as they continue to evolve over the term of any patent that issues herefrom. Applicant notes that an instructive earlier version of the AC-3 standard is described in detail in "Multi-Channel Digital Audio Compression System," Dolby Laboratories Information, Feb. 22, 1994.
Those of skill in the art will appreciate that the MPEG and AC-3 algorithms are well known, evolving standards. Accordingly, reference herein to these standards will be understood to mean the standards as they existed at the time of the earliest effective filing date of the present application, and as they have evolved to date, and as they continue to evolve over the term of any patent that issues herefrom. Applicant notes that an instructive earlier version of the AC-3 standard is described in detail in "Multi-Channel Digital Audio Compression System," Dolby Laboratories normation, Feb. 22, 1994.
The MPEG compression algorithm is the first international audio compression standard. According to the MPEG standard, effective compression can be obtained utilizing the human psychoacoustic recognition characteristic which responds differently depending on the frequency band. The AC-3 standard was adopted as the audio standard for North American High-Definition Television (HDTV) systems. The AC-3 standard has recently been applied to Digital Video Disk (DVD), Direct Broadcasting System (DBS), Set Top Box (STB), digital cable, etc. The AC-3 compression algorithm also uses the human psychoacoustic characteristic as a basis for audio compression. Both the MPEG and AC-3 standards are not limited to specific types of input signals and thus can be used for compressing speech, high-quality audio signals, and the like.
Recently, dual-mode audio decoders capable of decoding both the AC-3 audio stream and MPEG audio stream have been designed and introduced into the marketplace. To achieve such dual-mode audio decoders, it is necessary to unify the hardware blocks of the two audio standards. An audio decoder may be divided into a bit-allocation component and a reconstruction filter component for restoring a time-domain signal. Practically, the bit-allocation component for the MPEG standard is quite different from the bit allocation component of the AC-3 standard. Thus it is almost impossible to design bit allocation blocks having the same function as the MPEG-specific and AC-3-specific bit-allocation components. In contrast, the reconstruction filter components have similar functional blocks including inverse transform blocks, window blocks, and overlap and add blocks. The inverse transform blocks of MPEG and AC-3 are particularly suited for combination by properly modifying different transform equations adopted in the MPEG and AC-3 standards. Specifically, the MPEG and AC-3 standards adopt a subband structure which is efficient in processing audio signals. The subband structure of the MPEG and AC-3 standards are discussed in detail in P. P. Vaidyanathan, MULTIRATE SYSTEMS AND FILTER BANKS, Prentice Hall (1993) which is incorporated herein by reference. The frequency characteristic of each subband is expressed by a simple transform equation termed IMDCT. The IMDCT transform is discussed in further detail in J. P. Prinven and A. B. Bradley, Analysis/synthesis filter bank design based on time domain aliasing cancellation, IEEE Trans. Assp-34, Vol. No. 5, 1153-61 (Oct. 1986). The AC-3 standard supports three kinds of transform equations. One of the three transform equations is selected at the encoding stage according to the input signal characteristics. Thereafter, the selected transform equation is manipulated so that the FFT structure reduces the amount of computation. The MPEG standard, on the other hand, uses one transform equation which is different from the three types of AC-3 transform equations. Although the IMDCT of the MPEG standard is different from the IMDCT of the AC-3 standard, the IMDCT of the MPEG standard becomes the subset of the IMDCT of the AC-3 standard when the FFT is used. Accordingly, it is preferable to implement a dual-mode audio decoder which has an IMDCT circuit based on the same FFT structure. By doing so, the FFT structure can be shared by the MPEG and AC-3 specific components thereby reducing the overall decoder cost.
At the same time, a conventional IMDCT windowing method outputting MPEG data can be used with a more efficient memory structure since all 64 IMDCT outputs need not be simultaneously stored in memory.